From 9ab782d3943fd19cdce64af067a0974df2adcc65 Mon Sep 17 00:00:00 2001
From: Marek Blok <Marek.Blok@pg.edu.pl>
Date: Fri, 9 Apr 2021 11:51:57 +0200
Subject: [PATCH] Improvements + EN
---
.gitignore | 10 +-
Cw1/.vscode/launch.json | 6 +-
Cw1/.vscode/tasks.json | 6 +-
Cw1/Cw1_zad3.cpp | 129 --------
Cw1/{Cw1_zad3.cbp => Ex1_task3.cbp} | 6 +-
Cw1/Ex1_task3.cpp | 125 ++++++++
Cw1/Makefile | 6 +-
Cw1/Makefile.main | 2 +-
Cw1/{zad_1_syg_1.wav => ex_1_sig_1.wav} | Bin
Cw1/{zad_1_syg_2.wav => ex_1_sig_2.wav} | Bin
Cw1/matlab/design_task_2.m | 28 ++
Cw1/matlab/design_zad_2.m | 28 --
Cw1/matlab/{cw1_zad2.coef => ex1_task2.coef} | Bin
Cw1/matlab/{fileread.m => readaudiofile.m} | 16 +-
Cw1/matlab/readme.txt | 4 +-
Cw1/matlab/save_filter_coef.m | 302 ++++++++++++-------
Cw1/rundot.bat | 4 +-
17 files changed, 371 insertions(+), 301 deletions(-)
delete mode 100644 Cw1/Cw1_zad3.cpp
rename Cw1/{Cw1_zad3.cbp => Ex1_task3.cbp} (87%)
create mode 100644 Cw1/Ex1_task3.cpp
rename Cw1/{zad_1_syg_1.wav => ex_1_sig_1.wav} (100%)
rename Cw1/{zad_1_syg_2.wav => ex_1_sig_2.wav} (100%)
create mode 100644 Cw1/matlab/design_task_2.m
delete mode 100644 Cw1/matlab/design_zad_2.m
rename Cw1/matlab/{cw1_zad2.coef => ex1_task2.coef} (100%)
rename Cw1/matlab/{fileread.m => readaudiofile.m} (93%)
diff --git a/.gitignore b/.gitignore
index d6ba61b..de23a0c 100644
--- a/.gitignore
+++ b/.gitignore
@@ -1,8 +1,8 @@
-Cw1/Cw1_zad3_dbg.exe
-Cw1/Cw1_zad3_rls.exe
-Cw1/Cw1_zad3.dot
-Cw1/Cw1_zad3.gif
-Cw1/cw1_zad3.wav
+Cw1/Ex1_task3_dbg.exe
+Cw1/Ex1_task3_rls.exe
+Cw1/Ex1_task3.dot
+Cw1/Ex1_task3.gif
+Cw1/ex1_task3.wav
Cw1/log_file.txt
Cw1/out_win_dbg
Cw1/out_win_rls
diff --git a/Cw1/.vscode/launch.json b/Cw1/.vscode/launch.json
index 70221b6..c36eb40 100644
--- a/Cw1/.vscode/launch.json
+++ b/Cw1/.vscode/launch.json
@@ -5,10 +5,10 @@
"version": "0.2.0",
"configurations": [
{
- "name": "(gdb) Cw1 â debug run ",
+ "name": "(gdb) Ex1 â debug run ",
"type": "cppdbg",
"request": "launch",
- "program": "${workspaceFolder}/Cw1_zad3_dbg.exe",
+ "program": "${workspaceFolder}/Ex1_task3_dbg.exe",
"args": [],
"stopAtEntry": false,
"cwd": "${workspaceFolder}",
@@ -23,7 +23,7 @@
"ignoreFailures": true
}
],
- "preLaunchTask": "Build Cw1"
+ "preLaunchTask": "Build Ex1_task3.cpp"
},
{
"name": "(gdb) Launch",
diff --git a/Cw1/.vscode/tasks.json b/Cw1/.vscode/tasks.json
index 87f7221..48a340b 100644
--- a/Cw1/.vscode/tasks.json
+++ b/Cw1/.vscode/tasks.json
@@ -6,9 +6,9 @@
// ${command:cpptools.activeConfigName}
"tasks": [
{
- "label": "Build Cw1",
+ "label": "Build Ex1_task3.cpp",
"type": "shell",
- "command": "make build -f Makefile.main VS_CFG=${command:cpptools.activeConfigName}",
+ "command": "make build -f Makefile.main FILE=Ex1_task3 VS_CFG=${command:cpptools.activeConfigName}",
"group": {
"kind": "build",
"isDefault": true
@@ -19,7 +19,7 @@
"problemMatcher": "$gcc"
},
{
- "label": "Clean Cw1",
+ "label": "Clean Ex1_task3.cpp",
"type": "shell",
"command": "make clean -f Makefile.main VS_CFG=${command:cpptools.activeConfigName}",
"group": "build",
diff --git a/Cw1/Cw1_zad3.cpp b/Cw1/Cw1_zad3.cpp
deleted file mode 100644
index 1c4986d..0000000
--- a/Cw1/Cw1_zad3.cpp
+++ /dev/null
@@ -1,129 +0,0 @@
-/*! Laboratorium: Zaawansowane przetwarzanie sygnaĹĂłw telekomunikacji cyfrowej
- * Äw. 1. zad. 3.
- * - wczytanie z pliku odpowiedzi impulsowej filtru
- * (skrypt generujÄ
cy plik danych: ./matlab/design_zad_2.m
- * - klasyczny interpolator L = 2
- *
- * \author Marek Blok
- * \date 2018.03.01
- */
-#include <DSP_lib.h>
-
-int main(int argn, char *args[])
-{
- /*************************************************************/
- // Log file setup
- DSP::log.SetLogFileName("log_file.txt");
- DSP::log.SetLogState(DSP::e::LogState::file | DSP::e::LogState::console);
-
- DSP::log << DSP::lib_version_string() << std::endl << std::endl;
- /*************************************************************/
-
- DSP::LoadCoef coef_info;
- int N_LPF;
- DSP::Float_vector h_LPF;
-
- // wczytanie wspó³czynników z pliku matlab/cw1_zad2.coef
- coef_info.Open("cw1_zad2.coef", "matlab");
- N_LPF = coef_info.GetSize(0); // odczytanie liczby wspó³czynników pierwszeego wektora (indeksowanie od zera)
- if (N_LPF < 1)
- {
- DSP::log << DSP::e::LogMode::Error << "No filter coefficients: aboarding" << std::endl;
- return -1;
- }
- else
- {
- // wczytanie N_LPF wspó³czynników do tablicy h_LPF
- coef_info.Load(h_LPF);
- }
- /*************************************************************/
-
- DSP::Clock_ptr MasterClock;
- // Pozyskanie wskazania do g³ównego zegara algorytmu
- MasterClock=DSP::Clock::CreateMasterClock();
-
- long int Fp1, Fp2;
-
- // Ÿród³o próbek: plik ./test.wav (bie¿šcy katalog)
- DSP::u::WaveInput AudioIn(MasterClock, "test.wav", ".");
- // odczytanie szybkoĹci prĂłbkowania pliku powišzanego z bloczkiem AudioIn
- Fp1 = AudioIn.GetSamplingRate();
-
- Fp2 = 2*Fp1;
- // dwukrotny zeroinserter z wejĹciem pracujšcym w rytm zegara MasterClock
- DSP::u::Zeroinserter Zeroinserter(MasterClock, 2U);
- // przykÂład dodania tekstu do domyslnej nazwy bloczka (dla pliku *.dot)
- Zeroinserter.SetName("x2");
-
- // filtr typu FIR o rzeczywistych wspó³czynnikach odpowiedzi impulsowej (tutaj filtr interpolacyjny)
- // dÂługoĹĂŚ odpowiedzi impulsowej N_LPF, prĂłbki odpowiedzi impulsowej w tablicy h_LPF
- DSP::u::FIR InterpFIR(h_LPF);
-
- // wyjĹcie na kartĂŞ dŸwiĂŞkowš - szybkoĹĂŚ prĂłbkowania Fp2 (mono/16bit)
- DSP::u::AudioOutput SoundOut(Fp2, 1, 16);
- // zapis do pliku *.wav (mono/16bit, szybkosĂŚ prĂłbkowania Fp2)
- DSP::u::FileOutput FileOut("cw1_zad3.wav", DSP::e::SampleType::ST_short, 1, DSP::e::FileType::FT_wav, Fp2);
-
- /*************************************************************/
- // Definicje po³šczeù pomiêdzy bloczkami
- // wyjĹcie "out" bloczka AudioIn ³šczymy w wejĹciem "in" bloczka Zeroinserter
- AudioIn.Output("out") >> Zeroinserter.Input("in");
- // wyjĹcie "out" bloczka Zeroinserter ³šczymy w wejĹciem "in" bloczka InterpFIR
- Zeroinserter.Output("out") >> InterpFIR.Input("in");
- // wyjĹcie "out" bloczka InterpFIR ³šczymy w wejĹciem "in" bloczka SoundOut
- InterpFIR.Output("out") >> SoundOut.Input("in");
- // wyjĹcie "out" bloczka InterpFIR ³šczymy w wejĹciem "in" bloczka FileOut
- // Uwaga: jedno wyjĹcie moÂżna pod³šczyĂŚ do kilku wejĹĂŚ ale nie moÂżna
- // pod³šczyĂŚ kilku wyjĹĂŚ do jednego wejĹcia
- InterpFIR.Output("out") >> FileOut.Input("in");
-
-
- /////////////////////////////////
- // sprawdzenie, czy do wszystkich wejsÌ, wszystkich bloczków pod³šczone sš sygna³y wejsciowe
- DSP::Component::CheckInputsOfAllComponents();
-
- // *********************************** //
- // Zapis schematu zaimplementowanego algorytmu do pliku *.dot
- // (przetworzenie do pliku *.gif skryptem rundot.bat)
- DSP::Clock::SchemeToDOTfile(MasterClock, "Cw1_zad3.dot");
-
- // *********************************** //
- // przetwarzany w blokach po SamplesInSegment prĂłbek wejsciowych
- int SamplesInSegment = 512;
-
- // licznik przetworzonych prĂłbek wejsciowych
- __int64 NoOfSamplesProcessed = 0;
-
- // Maksymalna liczba przetwarzanych prĂłbek wejsciowych (10 sekund)
- #define MAX_SAMPLES_TO_PROCESS 10*Fp1
-
- while(NoOfSamplesProcessed < MAX_SAMPLES_TO_PROCESS)
- {
-
- // ********************************************************** //
- // uruchom zaimplementowany algorytm na SamplesInSegment cykli zegara MasterClock
- DSP::Clock::Execute(MasterClock, SamplesInSegment);
- // ********************************************************** //
-
- if (AudioIn.GetBytesRead() > 0)
- { // resetuj licznik przetworzonych prĂłbek jeÂżeli udaÂło siĂŞ odczytaĂŚ prĂłbki z pliku
- // - wymusza przetworzenie caÂłego pliku wejsciowego
- NoOfSamplesProcessed = 0;
- }
- else // Play 200ms more
- { // jeÂżeli nie odczytano prĂłbek z pliku wejsciowego to przetwĂłrz jeszcze tylko 1/5 sekundy
- if (NoOfSamplesProcessed < MAX_SAMPLES_TO_PROCESS - Fp1/5)
- NoOfSamplesProcessed = MAX_SAMPLES_TO_PROCESS - Fp1/5;
- }
-
- NoOfSamplesProcessed += SamplesInSegment;
- // ********************************************************** //
- }
-
- /*************************************************************/
- // Zwolnienie zarezerwowanych zasobĂłw
- DSP::Clock::FreeClocks();
- /*************************************************************/
-
- return 0;
-}
diff --git a/Cw1/Cw1_zad3.cbp b/Cw1/Ex1_task3.cbp
similarity index 87%
rename from Cw1/Cw1_zad3.cbp
rename to Cw1/Ex1_task3.cbp
index 4d61cab..37f8ce3 100644
--- a/Cw1/Cw1_zad3.cbp
+++ b/Cw1/Ex1_task3.cbp
@@ -2,12 +2,12 @@
<CodeBlocks_project_file>
<FileVersion major="1" minor="6" />
<Project>
- <Option title="Cw1_zad3" />
+ <Option title="Ex1_task3" />
<Option pch_mode="2" />
<Option compiler="gcc" />
<Build>
<Target title="default">
- <Option output="Cw1_zad3" prefix_auto="1" extension_auto="1" />
+ <Option output="Ex1_task3" prefix_auto="1" extension_auto="1" />
<Option type="1" />
<Option compiler="gcc" />
</Target>
@@ -27,7 +27,7 @@
<Add library="winmm" />
<Add directory="../DSPElib/CodeBlocks-m32_TDM_5.1.0/dbg" />
</Linker>
- <Unit filename="Cw1_zad3.cpp" />
+ <Unit filename="Ex1_task3.cpp" />
<Extensions>
<code_completion />
<envvars />
diff --git a/Cw1/Ex1_task3.cpp b/Cw1/Ex1_task3.cpp
new file mode 100644
index 0000000..17321af
--- /dev/null
+++ b/Cw1/Ex1_task3.cpp
@@ -0,0 +1,125 @@
+/*! Laboratory: Advanced signal processing of telecommunication signals
+ * (Zaawansowane przetwarzanie sygnaĹĂłw telekomunikacji cyfrowej)
+ * Ex. 1. task 3.
+ * - impulse response loaded from file
+ * (data generated with script: ./matlab/design_task_2.m)
+ * - classical interpolator L = 2
+ *
+ * \author Marek Blok
+ * \date 2021.04.09
+ */
+#include <DSP_lib.h>
+
+int main(int argn, char *args[])
+{
+ /*************************************************************/
+ // Log file setup
+ DSP::log.SetLogFileName("log_file.txt");
+ DSP::log.SetLogState(DSP::e::LogState::file | DSP::e::LogState::console);
+
+ DSP::log << DSP::lib_version_string() << std::endl << std::endl;
+ /*************************************************************/
+
+ DSP::LoadCoef coef_info;
+ int N_LPF;
+ DSP::Float_vector h_LPF;
+
+ // loading coefficients from file matlab/ex1_task2.coef
+ coef_info.Open("ex1_task2.coef", "matlab");
+ N_LPF = coef_info.GetSize(0); // read number of coefficients of first vector (index starts form zero)
+ if (N_LPF < 1)
+ {
+ DSP::log << DSP::e::LogMode::Error << "No filter coefficients: aborting" << std::endl;
+ return -1;
+ }
+ else
+ {
+ // Load coefficients to h_LPF vector
+ coef_info.Load(h_LPF);
+ }
+ /*************************************************************/
+
+ DSP::Clock_ptr MasterClock;
+ // Get pointer to algorithm's master clock
+ MasterClock=DSP::Clock::CreateMasterClock();
+
+ long int Fp1, Fp2;
+
+ // Source of samples: file ./DSPElib.wav (current folder)
+ DSP::u::WaveInput AudioIn(MasterClock, "DSPElib.wav", ".");
+ // get sampling rate associated with block AudioIn
+ Fp1 = AudioIn.GetSamplingRate();
+
+ Fp2 = 2*Fp1;
+ // two-fold zeroinserter with input working at rate of MasterClock
+ DSP::u::Zeroinserter Zeroinserter(MasterClock, 2U);
+ // example of appending text to the default block's name (for file *.dot)
+ Zeroinserter.SetName("x2");
+
+ // FIR filter with real valued impulse response coefficients (here an interpolation filter)
+ // uses samples from vector h_LPF
+ DSP::u::FIR InterpFIR(h_LPF);
+
+ // output to the soundcard - sampling rate Fp2 (mono/16bit)
+ DSP::u::AudioOutput SoundOut(Fp2, 1, 16);
+ // write to file *.wav (mono/16bit, sampling rate Fp2)
+ DSP::u::FileOutput FileOut("ex1_task3.wav", DSP::e::SampleType::ST_short, 1, DSP::e::FileType::FT_wav, Fp2);
+
+ /*************************************************************/
+ // Blocks connection definitions
+ // connecting output "out" of block AudioIn with input "in" of Zeroinserter block
+ AudioIn.Output("out") >> Zeroinserter.Input("in");
+ // connecting output "out" of Zeroinserter to input "in" of InterpFIR
+ Zeroinserter.Output("out") >> InterpFIR.Input("in");
+ // connecting output "out" of InterpFIR to input "in" of SoundOut
+ InterpFIR.Output("out") >> SoundOut.Input("in");
+ // connecting output "out" of InterpFIR to input "in" of FileOut
+ InterpFIR.Output("out") >> FileOut.Input("in");
+ // Note: one output can be connected to many inputs but many outputs cannot be conneted to one input
+
+
+ /////////////////////////////////
+ // Checking if all inputs of all blocks have attached input signals
+ DSP::Component::CheckInputsOfAllComponents();
+
+ // *********************************** //
+ // Saving the scheme of the implemented algorithm to the * .dot file
+ // (this file can be converted to *.gif using script rundot.bat
+ // adapt paths in batch file if necessary)
+ DSP::Clock::SchemeToDOTfile(MasterClock, "Ex1_task3.dot");
+
+ // *********************************** //
+ // processing in batches of SamplesInSegment input samples
+ int SamplesInSegment = 512;
+
+ // counter storing number of input samples processed
+ int64_t NoOfSamplesProcessed = 0;
+
+ // Maksimum number of input samples to process (200 miliseconds)
+ #define MAX_SAMPLES_TO_PROCESS Fp1/5
+
+ while(NoOfSamplesProcessed < MAX_SAMPLES_TO_PROCESS)
+ {
+
+ // ********************************************************** //
+ // run the implemented algorithm for the SamplesInSegment of the MasterClock clock cycles
+ DSP::Clock::Execute(MasterClock, SamplesInSegment);
+ // ********************************************************** //
+
+ if (AudioIn.GetBytesRead() > 0)
+ { // reset processed samples counter on successful input file read
+ // - forces processing of the whole input file
+ NoOfSamplesProcessed = 0;
+ }
+
+ NoOfSamplesProcessed += SamplesInSegment;
+ // ********************************************************** //
+ }
+
+ /*************************************************************/
+ // Releasing reserved resources
+ DSP::Clock::FreeClocks();
+ /*************************************************************/
+
+ return 0;
+}
diff --git a/Cw1/Makefile b/Cw1/Makefile
index c1b0952..c94c79f 100644
--- a/Cw1/Makefile
+++ b/Cw1/Makefile
@@ -18,19 +18,19 @@ ifeq ($(MODE),Release)
LINKER_FLAGS = $(comflag) -s -static-libgcc -static-libstdc++ $(MISC_LINKER_FLAGS)
INCLUDES := -I"$(DSPElib_DIR)/include" -I"$(DSPElib_DIR)/include/rls"
DSPElib_FULLDIR = $(DSPElib_DIR)/rls
- EXE_FILENAME = Cw1_zad3_rls.exe
+ EXE_FILENAME = $(CPP_FILENAME)_rls.exe
else
CFLAGS = $(comflag) -std=c++0x -O0 -g3 -Wall -c -fmessage-length=0 -W -Wshadow -Wconversion -fstrict-aliasing -fmax-errors=5
LINKER_FLAGS = $(comflag) -static-libgcc -static-libstdc++ $(MISC_LINKER_FLAGS)
INCLUDES := -I"$(DSPElib_DIR)/include" -I"$(DSPElib_DIR)/include/dbg"
DSPElib_FULLDIR = $(DSPElib_DIR)/dbg
- EXE_FILENAME = Cw1_zad3_dbg.exe
+ EXE_FILENAME = $(CPP_FILENAME)_dbg.exe
endif
# -U__STRICT_ANSI__ jest potrzebne do kompilacji debug_new.cpp, jezeli pominac ten plik to mozna rowniez wyrzucic te opcje
#CFLAGS_debug = $(comflag) -std=c++0x -O0 -g3 -Wall -c -fmessage-length=0 -W -Wshadow -Wco#nversion -fstrict-aliasing -U__STRICT_ANSI__
SOURCES_NAMES =
-SOURCES_NAMES += Cw1_zad3.cpp
+SOURCES_NAMES += $(CPP_FILENAME).cpp
SOURCES = $(addprefix $(SRC_CPP_SUBDIR)/,$(SOURCES_NAMES))
SOURCES_DBG =
diff --git a/Cw1/Makefile.main b/Cw1/Makefile.main
index cb3fad5..cf6c368 100644
--- a/Cw1/Makefile.main
+++ b/Cw1/Makefile.main
@@ -50,7 +50,7 @@ endif
build:
@echo "Building $(VS_CFG)"
@echo $(VS_CFG): $(MODE) // $(MAKEFILE)
- make build MODE=$(MODE) COMFLAG=$(COMFLAG) DFLAGS="$(DFLAGS)" LIBS="$(LIBS)" OUT_DIR=$(OUT_DIR) DSPElib_SUBDIR=$(DSPElib_SUBDIR) MISC_LINKER_FLAGS="$(MISC_LINKER_FLAGS)" -f $(MAKEFILE)
+ make build CPP_FILENAME=$(FILE) MODE=$(MODE) COMFLAG=$(COMFLAG) DFLAGS="$(DFLAGS)" LIBS="$(LIBS)" OUT_DIR=$(OUT_DIR) DSPElib_SUBDIR=$(DSPElib_SUBDIR) MISC_LINKER_FLAGS="$(MISC_LINKER_FLAGS)" -f $(MAKEFILE)
clean:
@echo "Cleaning $(VS_CFG)"
diff --git a/Cw1/zad_1_syg_1.wav b/Cw1/ex_1_sig_1.wav
similarity index 100%
rename from Cw1/zad_1_syg_1.wav
rename to Cw1/ex_1_sig_1.wav
diff --git a/Cw1/zad_1_syg_2.wav b/Cw1/ex_1_sig_2.wav
similarity index 100%
rename from Cw1/zad_1_syg_2.wav
rename to Cw1/ex_1_sig_2.wav
diff --git a/Cw1/matlab/design_task_2.m b/Cw1/matlab/design_task_2.m
new file mode 100644
index 0000000..aca9338
--- /dev/null
+++ b/Cw1/matlab/design_task_2.m
@@ -0,0 +1,28 @@
+function h=design_task_2
+
+%interpolation filter
+Fp1 = 22050; % input sampling rate
+Fp2 = 44100; % output sampling rate
+L = 2; % interpolation ratio (Fp2/Fp1)
+
+% passband ripples +/-0.1 dB, out of band attenuation -96dB
+dp = 10.^(0.1/20)-1;
+ds = 10.^(-96/20);
+
+% transition band 10-12kHz, filter works at rate equal to Fp2
+c = firpmord( [10000 12000], [1 0], [dp ds], Fp2, 'cell');
+h = firpm(c{:});
+N= length(h)
+
+figure(1)
+plot(h)
+% pause
+
+figure(2)
+freqz(h,1, 8*2048, L*Fp1)
+
+% writing filter coefficients to file
+dane.h = L*h;
+dane.Fp = Fp1;
+save_filter_coef('ex1_task2', dane);
+
diff --git a/Cw1/matlab/design_zad_2.m b/Cw1/matlab/design_zad_2.m
deleted file mode 100644
index d489173..0000000
--- a/Cw1/matlab/design_zad_2.m
+++ /dev/null
@@ -1,28 +0,0 @@
-function h=design_zad_2
-
-%interpolation filter
-Fp1 = 22050; % wejciowa szybkoć próbkowania
-Fp2 = 44100; % wyjciowa szybkoć próbkowania
-L = 2; % krotnoć interpolacji (Fp2/Fp1)
-
-% zafalowanie w pasmie +/-0.1 dB, tłumienie poza pasmem -96dB
-dp = 10.^(0.1/20)-1;
-ds = 10.^(-96/20);
-
-% Pasmo przejciowe 10-12kHz, filtr pracujšcy na szybkoci Fp2
-c = firpmord( [10000 12000], [1 0], [dp ds], Fp2, 'cell');
-h = firpm(c{:});
-N= length(h)
-
-figure(1)
-plot(h)
-% pause
-
-figure(2)
-freqz(h,1, 8*2048, L*Fp1)
-
-% zapisanie współczynników filtru do pliku
-dane.h = L*h;
-dane.Fp = Fp1;
-save_filter_coef('cw1_zad2', dane);
-
diff --git a/Cw1/matlab/cw1_zad2.coef b/Cw1/matlab/ex1_task2.coef
similarity index 100%
rename from Cw1/matlab/cw1_zad2.coef
rename to Cw1/matlab/ex1_task2.coef
diff --git a/Cw1/matlab/fileread.m b/Cw1/matlab/readaudiofile.m
similarity index 93%
rename from Cw1/matlab/fileread.m
rename to Cw1/matlab/readaudiofile.m
index 850c7b1..3a87be2 100644
--- a/Cw1/matlab/fileread.m
+++ b/Cw1/matlab/readaudiofile.m
@@ -1,5 +1,5 @@
-function [x, Fs] = fileread(filename, param)
-% [x, Fs] = fileread(filename, param)
+function [x, Fs] = readaudiofile(filename, param)
+% [x, Fs] = audiofileread(filename, param)
%
% returns vector x of the size SIZE=[samples channels].
% eg. x = x(:,1) + j*x(:,2);
@@ -12,7 +12,7 @@ function [x, Fs] = fileread(filename, param)
% *.flt
% *.wav
% *.tape
-% last modification: 2018.03.05
+% last modification: 2021.03.29
% Author: Marek Blok
return_cplx = 0;
@@ -46,22 +46,26 @@ end
switch file_type,
case 'w',
if strcmp(param, 'size') == 1,
- if exist('audioread','file') ~= 2
+ if exist('audioread','file') == 0
x = wavread(filename, 'size');
% siz = [samples channels].
else
info = audioinfo(filename);
x = [info.TotalSamples, info.NumChannels];
+ Fs = info.SampleRate;
end
else
if isfinite(param)
- if exist('audioread','file') ~= 2
+ if exist('audioread','file') == 0
[x, Fs] = wavread(filename, param);
else
+ if length(param) == 1
+ param = [1, param];
+ end
[x, Fs] = audioread(filename, param);
end
else
- if exist('audioread','file') ~= 2
+ if exist('audioread','file') == 0
[x, Fs] = wavread(filename);
else
[x, Fs] = audioread(filename);
diff --git a/Cw1/matlab/readme.txt b/Cw1/matlab/readme.txt
index 143af97..30b97d5 100644
--- a/Cw1/matlab/readme.txt
+++ b/Cw1/matlab/readme.txt
@@ -1,2 +1,2 @@
-wavread('../cw1_zad3.wav')
-fileread('../cw1_zad3.wav')
\ No newline at end of file
+audioread('../ex1_task3.wav')
+readaudiofile('../ex1_task3.wav')
\ No newline at end of file
diff --git a/Cw1/matlab/save_filter_coef.m b/Cw1/matlab/save_filter_coef.m
index 6931fa1..8cf1f07 100644
--- a/Cw1/matlab/save_filter_coef.m
+++ b/Cw1/matlab/save_filter_coef.m
@@ -10,147 +10,217 @@ function save_filter_coef(filename, coefficients, file_version)
% coefficients.a
% coefficients.b
%
-% For table of vectors see FIR filter coefficients above.
-%
% For file_version >= 1 (default == 0),
% coefficients.Fp - sampling frequency
%
% This file is a part of Digital Signal Processing Engine
+%
+% File format (*.coef) - this is open format, for general use
+% * (not only for storing coefficients)
+%
+% File format (*.h) - this is C++ header file for DSPElib with hardcoded *.coef equivalent
+%
% \author Marek Blok
-% \date 2008.03.19
+% \date 2020.02.22
if nargin == 2,
file_version = 0;
end
-% /*! File format (*.coef) - this is open format, for general use
-% * (not only for storing coefficients)
-% *
ind = find(filename == '.');
if length(ind) == 0,
filename = [filename, '.coef'];
end
-plik = fopen(filename, 'wb');
-% * - (uchar) 1B - file version number
-fwrite(plik, file_version, 'uchar')
+% detect mode based on file extention
+[PATHSTR,NAME,EXT] = fileparts(filename);
+switch EXT
+ case '.h'
+ mode = 'h-file';
+ plik = fopen(filename, 'wt');
-switch file_version,
- case 0,
- case 1,
- % * - (uint) 4B - Sampling frequency
- if isfield(coefficients, 'Fp'),
- fwrite(plik, coefficients.Fp, 'uint32')
- else
- fclose(plik);
- error('Input data does not contain Fp');
- end
- otherwise,
- fclose(plik);
- error('This version of coefficients file is unsupported');
+ case '.coef'
+ mode = 'coefs-file';
+ plik = fopen(filename, 'wb');
+
+ otherwise
+ mode = 'unknown-file';
+ error(mode);
end
-if isfield(coefficients, 'h'),
- isFIR = 1;
- if iscell(coefficients.h)
- resp_no = length(coefficients.h);
-% if resp_no = 1;
-% coefficients.h = coefficients.h{1};
-% end
- else
- resp_no = 1;
- end
- if resp_no == 1,
- isComplex = any(imag(coefficients.h(:)));
- else
- isComplex = false;
- for ind_resp = 1:resp_no,
- isComplex = isComplex | any(imag(coefficients.h{ind_resp}(:)));
+switch mode
+ case 'h-file'
+ fprintf(plik, '// save_filter_coef output (hard coded coefficients)\n');
+ fprintf(plik, '#pragma once\n');
+ fprintf(plik, '#include <DSP_lib.h>\n');
+ fprintf(plik, '\n');
+
+ if isfield(coefficients, 'h'),
+ h = coefficients.h;
+
+ for ind_h = 0:length(h)-1
+ fprintf(plik, 'DSP_float_vector Farrow_coefs_row_%i = {', ind_h);
+ h_tmp = h{ind_h+1};
+ for ind_p = 0:length(h_tmp)-1,
+ if ind_p > 0
+ fprintf(plik, ', ');
+ end
+ fprintf(plik, '%.15ff', h_tmp(ind_p+1));
+ end
+ %0.1f, 0.2f, 0.1f
+ fprintf(plik, '};\n');
+ end
+ fprintf(plik, '\n');
+
+ %vector <DSP_float_vector> Farrow_coefs = { coefs_0, coefs_1, coefs_2 };
+ fprintf(plik, 'vector <DSP_float_vector> Farrow_coefs = {');
+ for ind_h = 0:length(h)-1
+ if ind_h > 0
+ fprintf(plik, ', ');
+ end
+ if (ind_h < length(h_tmp)) & (rem(ind_h, 5) == 0)
+ fprintf(plik, '\n ');
+ end
+ fprintf(plik, 'Farrow_coefs_row_%i', ind_h);
+ end
+ fprintf(plik, '\n };\n');
+
+ fprintf(plik, '\n');
+ fprintf(plik, 'const unsigned int p_Farrow = %i; // should be equal to Farrow_coefs[0].size()-1\n', length(h{1})-1);
+ fprintf(plik, 'const unsigned int N_FSD = %i; // should be equal to Farrow_coefs.size()\n', length(h));
end
- end
-else
- isFIR = 0;
- isComplex = any(imag(coefficients.a(:))) | any(imag(coefficients.b(:)));
-end
+
+ if isfield(coefficients, 'b'),
+ fclose all;
+ error('unsupported: coefficients.b');
+ end
+ if isfield(coefficients, 'a'),
+ fclose all;
+ error('unsupported: coefficients.a');
+ end
+
+ case 'coefs-file'
+ % * - (uchar) 1B - file version number
+ fwrite(plik, file_version, 'uchar');
-% * - data - coefficients data (depends on fle version)
-% * .
-% * Data segment format:
-% * -# (version: 0x00)
-% * - (uchar) 1B - number of sample dimensions
-% * 1 - real, 2 - complex, ...
-if isComplex,
- fwrite(plik, 2, 'uchar'); % complex
-else
- fwrite(plik, 1, 'uchar'); % real
-end
-% * - (uchar) 1B - sample component type
-% * - DSP_FT_float (=1) : C++ float (32bit floating point)
-% * - DSP_FT_short (=2) : C++ short (16bit signed integer)
-% * - DSP_FT_uchar (=3) : C++ unsigned char (8bit unsigned integer with bias (0x80))
-% * - DSP_FT_double (=7) : C++ double (64bit floating point)
-% * - DSP_FT_long_double (=8) : C++ long double (80bit floating point)
-fwrite(plik, 1, 'uchar');
-
-% * - (uchar) 1B - number of vectors
-% * - 1 - FIR filter coefficients (one vector)
-% * - 2 - IIR filter coefficients (two vectors)
-% * - (x number of vectors)
-% * - (ushort) 2B - number of samples in vector
-% * - (x number of samples)
-% * - (x number of sample dimensions)
-% * - (sample componet type) xB - sample component
-% * e.g. real, imag part
-if isFIR,
- fwrite(plik, resp_no, 'uchar');
-
- if iscell(coefficients.h)
- for ind_resp = 1:resp_no,
- N_FIR = length(coefficients.h{ind_resp});
- fwrite(plik, N_FIR, 'uint16');
- if isComplex,
- dane(1:2:2*N_FIR) = real(coefficients.h{ind_resp});
- dane(2:2:2*N_FIR) = imag(coefficients.h{ind_resp});
- fwrite(plik, dane, 'float');
+ switch file_version,
+ case 0,
+ case 1,
+ % * - (uint) 4B - Sampling frequency
+ if isfield(coefficients, 'Fp'),
+ fwrite(plik, coefficients.Fp, 'uint32');
+ else
+ fclose(plik);
+ error('Input data does not contain Fp');
+ end
+ otherwise,
+ fclose(plik);
+ error('This version of coefficients file is unsupported');
+ end
+
+ if isfield(coefficients, 'h'),
+ isFIR = 1;
+ if iscell(coefficients.h)
+ resp_no = length(coefficients.h);
+ % if resp_no = 1;
+ % coefficients.h = coefficients.h{1};
+ % end
+ else
+ resp_no = 1;
+ end
+ if resp_no == 1,
+ isComplex = any(imag(coefficients.h(:)));
else
- fwrite(plik, real(coefficients.h{ind_resp}), 'float');
+ isComplex = false;
+ for ind_resp = 1:resp_no,
+ isComplex = isComplex | any(imag(coefficients.h{ind_resp}(:)));
+ end
end
+ else
+ isFIR = 0;
+ isComplex = any(imag(coefficients.a(:))) | any(imag(coefficients.b(:)));
end
- else
- N_FIR = length(coefficients.h);
- fwrite(plik, N_FIR, 'uint16');
+
+ % * - data - coefficients data (depends on fle version)
+ % * .
+ % * Data segment format:
+ % * -# (version: 0x00)
+ % * - (uchar) 1B - number of sample dimensions
+ % * 1 - real, 2 - complex, ...
if isComplex,
- dane(1:2:2*N_FIR) = real(coefficients.h);
- dane(2:2:2*N_FIR) = imag(coefficients.h);
- fwrite(plik, dane, 'float');
+ fwrite(plik, 2, 'uchar'); % complex
else
- fwrite(plik, real(coefficients.h), 'float');
+ fwrite(plik, 1, 'uchar'); % real
end
- end
-
-else
- fwrite(plik, 2, 'uchar');
-
- N_a = length(coefficients.a);
- fwrite(plik, N_a, 'uint16');
- if isComplex,
- dane(1:2:2*N_a) = real(coefficients.a);
- dane(2:2:2*N_a) = imag(coefficients.a);
- fwrite(plik, dane, 'float');
- else
- fwrite(plik, real(coefficients.a), 'float');
- end
-
-
- N_b = length(coefficients.b);
- fwrite(plik, N_b, 'uint16');
- if isComplex,
- dane(1:2:2*N_b) = real(coefficients.b);
- dane(2:2:2*N_b) = imag(coefficients.b);
- fwrite(plik, dane, 'float');
- else
- fwrite(plik, real(coefficients.b), 'float');
- end
-end
+ % * - (uchar) 1B - sample component type
+ % * - DSP_FT_float (=1) : C++ float (32bit floating point)
+ % * - DSP_FT_short (=2) : C++ short (16bit signed integer)
+ % * - DSP_FT_uchar (=3) : C++ unsigned char (8bit unsigned integer with bias (0x80))
+ % * - DSP_FT_double (=7) : C++ double (64bit floating point)
+ % * - DSP_FT_long_double (=8) : C++ long double (80bit floating point)
+ fwrite(plik, 1, 'uchar');
+
+ % * - (uchar) 1B - number of vectors
+ % * - 1 - FIR filter coefficients (one vector)
+ % * - 2 - IIR filter coefficients (two vectors)
+ % * - (x number of vectors)
+ % * - (ushort) 2B - number of samples in vector
+ % * - (x number of samples)
+ % * - (x number of sample dimensions)
+ % * - (sample componet type) xB - sample component
+ % * e.g. real, imag part
+ if isFIR,
+ fwrite(plik, resp_no, 'uchar');
+
+ if iscell(coefficients.h)
+ for ind_resp = 1:resp_no,
+ N_FIR = length(coefficients.h{ind_resp});
+ fwrite(plik, N_FIR, 'uint16');
+ if isComplex,
+ dane(1:2:2*N_FIR) = real(coefficients.h{ind_resp});
+ dane(2:2:2*N_FIR) = imag(coefficients.h{ind_resp});
+ fwrite(plik, dane, 'float');
+ else
+ fwrite(plik, real(coefficients.h{ind_resp}), 'float');
+ end
+ end
+ else
+ N_FIR = length(coefficients.h);
+ fwrite(plik, N_FIR, 'uint16');
+ if isComplex,
+ dane(1:2:2*N_FIR) = real(coefficients.h);
+ dane(2:2:2*N_FIR) = imag(coefficients.h);
+ fwrite(plik, dane, 'float');
+ else
+ fwrite(plik, real(coefficients.h), 'float');
+ end
+ end
+ else
+ fwrite(plik, 2, 'uchar');
+
+ N_a = length(coefficients.a);
+ fwrite(plik, N_a, 'uint16');
+ if isComplex,
+ dane(1:2:2*N_a) = real(coefficients.a);
+ dane(2:2:2*N_a) = imag(coefficients.a);
+ fwrite(plik, dane, 'float');
+ else
+ fwrite(plik, real(coefficients.a), 'float');
+ end
+
+
+ N_b = length(coefficients.b);
+ fwrite(plik, N_b, 'uint16');
+ if isComplex,
+ dane(1:2:2*N_b) = real(coefficients.b);
+ dane(2:2:2*N_b) = imag(coefficients.b);
+ fwrite(plik, dane, 'float');
+ else
+ fwrite(plik, real(coefficients.b), 'float');
+ end
+ end
+
+end
fclose(plik);
+
diff --git a/Cw1/rundot.bat b/Cw1/rundot.bat
index 1aacf51..5e24fef 100644
--- a/Cw1/rundot.bat
+++ b/Cw1/rundot.bat
@@ -1,4 +1,4 @@
path = "D:\Program Files (x86)\Graphviz\bin";%path%
-del Cw1_zad3.gif
-dot -Tgif Cw1_zad3.dot -oCw1_zad3.gif
+del Ex1_task3.gif
+dot -Tgif Ex1_task3.dot -oEx1_task3.gif
--
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